This patch adds res_srtp module using libsrtp to support SRTP in asterisk.
Signed-off-by: Victor Seva <linuxman...@torreviejawireless.org>
---
net/asterisk-1.8.x/Makefile | 5 +++--
1 files changed, 3 insertions(+), 2 deletions(-)
diff --git a/net/asterisk-1.8.x/Makefile b/net/asterisk-1.8.x/Makefile
index aa316ae..361fa22 100644
--- a/net/asterisk-1.8.x/Makefile
+++ b/net/asterisk-1.8.x/Makefile
@@ -39,7 +39,7 @@ define Package/asterisk18
$(call Package/asterisk18/Default)
TITLE:=Complete open source PBX, v1.8x
MENU:=1
- DEPENDS:= +libopenssl +libncurses +libpopt +libpthread +zlib @!TARGET_avr32
+ DEPENDS:= +libopenssl +libncurses +libpopt +libpthread +zlib +libsrtp
@!TARGET_avr32
endef
define Package/asterisk18/description
@@ -258,6 +258,7 @@ CONFIGURE_ARGS+= \
--without-tinfo \
--without-vorbis \
--without-vpb \
+ --with-srtp="$(STAGING_DIR)/usr" \
--with-z="$(STAGING_DIR)/usr" \
--with-sounds-cache="$(DL_DIR)" \
--disable-xmldoc
@@ -318,7 +319,7 @@ define Package/asterisk18/install
done
$(INSTALL_DIR) $(1)/usr/lib/asterisk/modules
for f in app_dial app_echo app_playback app_macro \
- chan_sip res_rtp_asterisk res_rtp_multicast \
+ chan_sip res_rtp_asterisk res_rtp_multicast res_srtp \
codec_ulaw codec_gsm \
format_gsm format_pcm format_wav format_wav_gsm \
pbx_config \
--
1.7.7.3
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